SDP Options - CS5000

[skipskip]

Hi Guys,

Is there a way to change the SPD options on a CS5000(V5.1.0.46), like on the Mitel 3300?


We're trying to troubleshoot a SIP Trunk problem where the SDP/SIP Headers are acting up.

http://britishschoolsofamerica.org/uwi/help/En/sysadmin/forms/sip_peers.html

Thanks

[Tech Electronics]

Skipskip,

What exactly is your problem?

Thanks,

TE

[skipskip]

Hi Tech,

We are having problem getting SIP to work from an AudioCodes 1000 to our CS5000. (See my other post. http://mitelforums.com/forum/index.php/topic,5037.0.html)

When making a SIP call...

- The CS5000 sends an Invite packet to the AC1000.
- The AC1000 sends 'Trying' packet , then 'Ringing'
- RTP is flowing from AC1000 to the CS5000, but to UDP ports that are not ready, as the 5000 is not happy that a call has started.
- The AC1000 processes the call, then sends back a '200 OK' packet when the call is answered at the remote end.
- The 5000 does not 'ACK' the '200 OK'.
- The AC1000 ends the call, due to no response from the CS5000.
- The 5000 responds with a '400 Not found' packet , indicating that there is no call to end anyway.

There seems to be a mismatch with the SDP between the Audio Codes Gateway and the CS5000. In doing research one of the fixes for this on the Mitel 3300 is to disable the Mitel proprietary SDP. However looking at the CS5000 I'm unable to find any SDP settings, hence my reason for asking.

Thanks

[Tech Electronics]

Skipskip,

First of all I do not believe the Audiocodes 1000 is a supported gateway for the 5000. That being said the issue most likely resides in the fact that the UDP ports that the 5000 uses are not the same as those for the Audiocodes unit.

Is the packet capture that you have on the other post from this configuration? If not could you recreate the problem with a new packet capture?

Addendum:
Alright, I have gone over your packet capture since I had a few minutes to kill and the first thing that jumps out at me is that the Audiocodes (Mediant) Gateway is sending to a user that does not exist within the Mitel 5000. Most likely this stems from the audiocodes not being programmed correctly.

Is this an MGCP Gateway or a SIP Gateway? It depends on which type it is on how to program the 5000 as well as the gateway.

Thanks,

TE

[skipskip]

Hi Tech,

Your correct about the AudioCodes Mediant 1000 is not support by Mitel. The Mediant is a SIP gateway. The reason for the Mediant though is because we will be delivering SIP to our Mitel CS5000 AND also a Lync enterprise voice environment. The Mediant allows the flexibility for us to reroute the SIP calls as we see fit.

If you have a look at the attached packet capture (This was taken today)

The INVITE packet is send from 10.31.1.3:5060 (Mitel Processor Card) to 10.31.1.4:5060 (Mediant 1000). In this packet in the L7 headers its saying to use 10.31.1.2:6846 (Mitel Processor Expansion Card) for RTP traffic.

Then when the first RTP packet is sent from 10.31.1.4:6340 (Mediant 1000) to the 10.31.1.2:6846 (Mitel Processor Expansion Card) a ICMP packet is send back saying Port Unreachable... So its like the PEC hasnt opened up port 6846, which was sent in the original INVITE packet.

Thanks

[skipskip]

Here is the Processor and Expansion Module config

[Tech Electronics]

Skipskip,

Could you post your configs for the Audiocodes, the Mitel ports are default and I need to see how your Gateway is configured as it is sending information in the wrong format.

Thanks,

TE

[skipskip]

Hi Teach, I only have read access to the Mediant as its managed by the SIP provider. Are you able to tell me what exactly your looking for and I'll try and get the config.

Thanks

[Tech Electronics]

Skipskip,

What is normally required on the Audiocodes devices is that the Hopoff Directory and Sip Trunks are set up properly so they work with the 5000. I will try and post a more formal answer once I get access to a device.

Update:

Here is the information that I believe is setup wrong in the Gateway.

Hopoff Number Directories
You must create a separate Hopoff Number Directory for each SIP trunk connected to the gateway. When placing an outbound call using a SIP trunk, the PBX sends the trunk extension number followed by the dialed digits to the SIP gateway. The Hopoff Number Directories are used to remove the trunk number before echoing digits to the CO. Each Hopoff Number Directory cannot have more than one number pattern (or replacement pair).

Trunk Circuit Routing Groups
In the Forced Routing Number Type box, type the SIP trunk number for forced routing number calls. You must do this or SIP invite messages will not have a destination number or they will have a phone number. The Forced Routing Number is basically the reverse of the Hopoff Number Directories. It adds the trunk number to incoming calls so the PBX knows which trunk is receiving the call. <--- They are sending you a {phone number}@IP Address which means nothing to the 5000.
Thanks,

TE